asterisk users - Oct 2009 - Audio issue in skype for asterisk

Hi,

I am facing audio issue in my skype for asterisk setup.

*Flow of the call is like this.*

e.g.
Skype users :
test2

Sip users:
1001
1002 <--> test2

This both sip users 1001 and 1002 are register in same asterisk. And also
test2 skype user is register in same asterisk.

Now 1001 is dialing skype user test2 (skypeout). So, test2 is getting call.
But as test2 skype user is register in our asterisk, our asterisk is getting
that call (skypein). And test2 is mapped with 1002 user. So when test2 user
call comes to asterisk our asterisk is dialing SIP/1002. And 1002 is getting
calls. But when 1001 and 1002 user is connecting they are not getting audio.
But this is working fine for only skypout and skypein. But when call come
back to asterisk audio issue is coming.

I have checked rtp debug, But getting proper packages in rtp debug.

I am attaching image of call flow.
[image:
?ui=2&view=att&th=1247ba299ad2be9d&attid=0.1&disp=attd&realattid=ii_1247ba299ad2be9d&zw]

Please help me to fix the issue.

-- 

Thanks,
Samir Doshi
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20091022/e044263a/attachment.htm
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 6004 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20091022/e044263a/attachment.jpeg

Hi,

we have a similar problem. When we try to make two skype-calls at a time,
only one of them has working audio. For this to happen, both calls must be
ringing at the same time. Does anyone know how to fix this?

Best regards,
Marcus Hunger

On Thu, Oct 22, 2009 at 10:45 AM, Samir Doshi <smrdoshi at gmail.com>
wrote:
> Hi,
>
> I am facing audio issue in my skype for asterisk setup.
>
> *Flow of the call is like this.*
>
> e.g.
> Skype users :
> test2
>
> Sip users:
> 1001
> 1002 <--> test2
>
> This both sip users 1001 and 1002 are register in same asterisk. And also
> test2 skype user is register in same asterisk.
>
> Now 1001 is dialing skype user test2 (skypeout). So, test2 is getting call.
> But as test2 skype user is register in our asterisk, our asterisk is
getting
> that call (skypein). And test2 is mapped with 1002 user. So when test2 user
> call comes to asterisk our asterisk is dialing SIP/1002. And 1002 is
getting
> calls. But when 1001 and 1002 user is connecting they are not getting
audio.
> But this is working fine for only skypout and skypein. But when call come
> back to asterisk audio issue is coming.
>
> I have checked rtp debug, But getting proper packages in rtp debug.
>
> I am attaching image of call flow.
> [image:
>
?ui=2&view=att&th=1247ba299ad2be9d&attid=0.1&disp=attd&realattid=ii_1247ba299ad2be9d&zw]
>
> Please help me to fix the issue.
>
> --
>
> Thanks,
> Samir Doshi
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
Dipl.-Inf. (FH)
Marcus Hunger - hunger at sipgate.de
Telefon: +49 (0)211-63 55 55-61
Telefax: +49 (0)211-63 55 55-22

sipgate GmbH - Gladbacher Str. 74 - 40219 D?sseldorf
HRB D?sseldorf 39841 - Gesch?ftsf?hrer: Thilo Salmon, Tim Mois
Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391

www.sipgate.de - www.sipgate.at - www.sipgate.co.uk
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20091130/852b27bd/attachment.htm
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 6004 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20091130/852b27bd/attachment.jpeg